1. Field of the Invention
The present invention relates to a Voice over Internet Protocol system and, more particularly, to a Voice over Internet Protocol system having a dynamic gain control function and a method for providing a dynamic gain using the system wherein in the process of converting Pulse Code Modulation (PCM) data to a Voice over Internet Protocol packet or vice versa, a dynamic gain value is assigned in accordance with the type of call, and PCM data can be amplified and outputted in accordance with the assigned gain value.
2. Description of the Related Art
Generally, communication apparatuses connected to telephone lines include a general telephone, a facsimile, and so on. A signal intensity of a communication apparatus connected to a telephone line is changed in accordance with the distance from an exchange. That is, since the telephone line made up of a general copper wire has impedance values changing depending upon its length, a loop current relative to a fixed direct current (DC) of the exchange also changes.
Since it may be said that the change of the loop current indicates the change of the length, the change of the signal according to the distance from the exchange is proportional to the change of the loop current. So, a communication equipment user separated far from the exchange hears the other party's voice as a very low sound or never hears it under certain circumstances.
Accordingly, it is necessary to compensate a signal on the attenuating telephone line according to its distance from the exchange. And, it is possible to have the other party's voice heard well even in communication equipment separated far from the exchange by preparing a gain control circuit in the exchange and controlling a gain of a signal received from a terminal.
FIG. 1 is a view showing a representative block construction of an exemplary subscriber terminal gain control circuit of an exchange in the art.
Referring to FIG. 1, the gain control circuit of the exchange subscriber terminal includes a subscriber connector 110 for connecting the exchange to a subscriber terminal, and an amplification part 120 for amplifying a signal provided from the subscriber connector 110 to a desired level.
A subscriber terminal (not shown) to be subscribed and connected to the exchange is connected to the exchange through the subscriber connector 110.
A voice signal applied from the terminal through the subscriber connector 110 is applied to and amplified in the amplification part 120. Here, an amplifier 126 decides an amplification gain by a ratio of a first resistor 122, an input resistor, and a second resistor 124, an output resistor, and the gain can be calculated according to the following expression.amplified gain=amplification gain ratio*20 Log (R2/R1)  (Expression 1)
Here, R1 and R2 indicate an input resistor and an output resistor respectively. As shown in expression 1, the amplification gain can be variable by properly controlling the ratio of the first resistor 122 of the input resistor and the second resistor 124 of the output resistor.
An analog voice signal amplified and applied from the amplification part 120 is applied to PCM converter 128 to be converted to a digital signal, and the converted digital signal is applied to a controller 130 so as to perform a process needed for exchanging.
However, in case of using the resistor 122 and the resistor 124 which have fixed values respectively, receiver sensitivity is transmitted as it is, regardless of the condition of a cable since fixed transmission and receiver gains are maintained despite the change of input state due to an external condition.
Since a call gain control circuit in the art has only a unidirectional gain value using fixed elements, it becomes difficult to control the transmission and receiver gains in accordance with characteristic of each line in the real situation that may have various states.
In order to solve this problem, a programmable gain control circuit has been developed wherein transmission and reception lines each have a number of resistors which are the same for each line in number and serially connected to each line and on/off switches are connected in parallel to the resistors so that the on/off of the switches are controlled by the controller in accordance with states of the cables and then the values of the resistors to determine the controlled transmission and receiver gains.
On the other hand, a Voice over Internet Protocol (referred to as a VoIP, hereinafter) is a communication service of new mode wherein a voice communication is performed not through an existing communication network, a Public Switch Telephone Network (referred to as a PSTN, hereinafter), but through an Internet network. Since the communication method using the Internet network uses a packet-based network different from existing communication methods, a user does not have to pay for charges of domestic/international phone lines separately so that it is possible to perform the voice communication at a lower fare.
The VoIP has a faculty of transmitting video information as well as audio information using an H.323 Protocol being an ITU-T (International Telecommunication Union—Telecommunication) standard that provides fundamental principles for voice, video, and data communications over the IP (Internet Protocol) network including the Internet. One of the H.323 entities defined in the H.323 protocol is a gatekeeper. The gatekeeper binds H.323 endpoints present in a packet-based network (i.e., an IP-based network) in one control zone defined as a “Zone”, and then controls/manages the bound H.323 endpoints.
A VoIP system using the Internet network as a back-bone has an exemplary construction as shown in FIG. 2.
Referring to FIG. 2, the VoIP system has an Internet 208 used as the back-bone, and the Internet 208 is connected to gateways 206 and 210 and to personal computers 216 and 218 (referred to as PCs, hereinafter).
The gateways 206 and 210 are correspondingly connected to PSTNs 204 and 212 that also are connected to telephone terminals 202 and 214 respectively. Terminals such as the phone terminals 202 and 214 and the PCs 216 and 218 are endpoints which are capable of communicating by voices (essential), images (option) and data (option) during a one-to-one communication or a conference.
Such terminals can perform a real-time and a bidirectional communication with the gateways 206 and 210 and other terminals. The gateways 206 and 210 are elements that enable terminals (for example, PCs 216 and 218) connected to the Internet 208 being a packet-based network and terminals (for example, telephone terminals 202 and 214) connected to the PSTNs 204 and 212 or an Integrated Service Digital Network (referred to as an ISDN, hereinafter) to perform the real-time and the bidirectional communication.
Briefly, gateways 206 and 210 perform a real-time compression and a protocol transformation of voices and facsimile data inputted from the PSTNs 204 and 212 and send the data to the Internet 208.
The Internet phone (IP-phone) can be classified into 3 types according to the kind of terminal used at both ends, that is, PC to PC, PC to phone and phone to phone.
Generally, the technical principle of the Internet phone is made up of a voice encoding and compressing technology, a real-time data transmission technology, a packet recovery technology, a gateway technology, and so on.
The voice encoding technology employs a low bit rate, high compression rate and high voice quality encoding technique in order to transmit the voice information without damaging the voice quality.
The voice encoding technology includes a PCM, an adaptive prediction coding, a Global System for Mobile communication (referred to as GSM, hereinafter), a Linear Predictive Coding (referred to as LPC, hereinafter), and so on, and the above technologies are now used. The real-time transformation technology includes a Real Time Transport Protocol (referred to as RTP, hereinafter). The RTP receives much recognition in the transmission quality over the Internet and is mainly used since 1995.
Also, a gateway embodying technology is to embody a gateway which is a network connection apparatus transforming analog voice information in order to transmit the information from an existing line exchange network to a packet exchange network.
A basic function of the gateway is processed in a digital signal processor. The gateway performs a voice compression capability using a compression algorithm, a waiting capability, and a removal capability so that it is possible to transform and transmit PCM voice data to a VoIP packet, and vice versa.
Here, in the process that the digital signal processor transforms the PCM data into the VoIP packet by performing the voice compression capability using the compression algorithm, the waiting capability, and the removal capability, the strength of voice may become too low or too high. Therefore, the gateway has a capability of controlling a gain value before compressing the PCM data to the VoIP packet, like the gain controlling circuit in the exchange described above.
However, since the gain value in the gateway is already assigned by the operator as a fixed value (of course, the gain value is generally determined by a test according to an environmental characteristic of the gateway), it is difficult to dynamically change the gain value according to the kind of call.